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gst/rtp/: Ported mulaw and alaw payloaders to use new base class
Original commit message from CVS: * gst/rtp/gstrtppcmapay.c: * gst/rtp/gstrtppcmapay.h: * gst/rtp/gstrtppcmupay.c: * gst/rtp/gstrtppcmupay.h: Ported mulaw and alaw payloaders to use new base class
This commit is contained in:
Philippe Kalaf
parent
c209a3f894
commit
1be3219c70
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@ -1,3 +1,10 @@
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2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
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* gst/rtp/gstrtppcmapay.c:
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* gst/rtp/gstrtppcmapay.h:
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* gst/rtp/gstrtppcmupay.c:
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* gst/rtp/gstrtppcmupay.h:
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Ported mulaw and alaw payloaders to use new base class
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2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
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* po/af.po:
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@ -59,17 +59,9 @@ static GstStaticPadTemplate gst_rtp_pcma_pay_src_template =
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static gboolean gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_pcma_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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static void gst_rtp_pcma_pay_finalize (GObject * object);
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GST_BOILERPLATE (GstRtpPmcaPay, gst_rtp_pcma_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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/* The lower limit for number of octet to put in one packet
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* (clock-rate=8000, octet-per-sample=1). The default 80 is equal
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* to to 10msec (see RFC3551) */
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#define GST_RTP_PCMA_MIN_PTIME_OCTETS 80
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GST_BOILERPLATE (GstRtpPmcaPay, gst_rtp_pcma_pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtp_pcma_pay_base_init (gpointer klass)
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@ -95,30 +87,24 @@ gst_rtp_pcma_pay_class_init (GstRtpPmcaPayClass * klass)
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_rtp_pcma_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_pcma_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_pcma_pay_handle_buffer;
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}
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static void
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gst_rtp_pcma_pay_init (GstRtpPmcaPay * rtppcmapay, GstRtpPmcaPayClass * klass)
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{
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rtppcmapay->adapter = gst_adapter_new ();
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmapay);
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GST_BASE_RTP_PAYLOAD (rtppcmapay)->clock_rate = 8000;
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}
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static void
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gst_rtp_pcma_pay_finalize (GObject * object)
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{
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GstRtpPmcaPay *rtppcmapay;
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/* tell basertpaudiopayload that this is a sample based codec */
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gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
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rtppcmapay = GST_RTP_PCMA_PAY (object);
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g_object_unref (rtppcmapay->adapter);
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rtppcmapay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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/* octet-per-sample is 1 for PCM */
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gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1);
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}
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static gboolean
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@ -132,114 +118,6 @@ gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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return TRUE;
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}
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static GstFlowReturn
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gst_rtp_pcma_pay_flush (GstRtpPmcaPay * rtppcmapay, guint32 clock_rate)
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{
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guint avail;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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guint maxptime_octets = G_MAXUINT;
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guint minptime_octets = GST_RTP_PCMA_MIN_PTIME_OCTETS;
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if (GST_BASE_RTP_PAYLOAD (rtppcmapay)->max_ptime > 0) {
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/* calculate octet count with:
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maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
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maxptime_octets =
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gst_util_uint64_scale_int (GST_BASE_RTP_PAYLOAD (rtppcmapay)->max_ptime,
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clock_rate, GST_SECOND);
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}
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/* the data available in the adapter is either smaller
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* than the MTU or bigger. In the case it is smaller, the complete
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* adapter contents can be put in one packet. */
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avail = gst_adapter_available (rtppcmapay->adapter);
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ret = GST_FLOW_OK;
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while (avail >= minptime_octets) {
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guint8 *payload;
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guint8 *data;
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guint payload_len;
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guint packet_len;
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/* fill one MTU or all available bytes */
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payload_len =
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MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmapay), maxptime_octets),
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avail);
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/* this will be the total lenght of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
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/* create buffer to hold the payload */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* copy payload */
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gst_rtp_buffer_set_payload_type (outbuf,
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GST_BASE_RTP_PAYLOAD_PT (rtppcmapay));
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payload = gst_rtp_buffer_get_payload (outbuf);
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data = (guint8 *) gst_adapter_peek (rtppcmapay->adapter, payload_len);
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memcpy (payload, data, payload_len);
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gst_adapter_flush (rtppcmapay->adapter, payload_len);
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avail -= payload_len;
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GST_BUFFER_TIMESTAMP (outbuf) = rtppcmapay->first_ts;
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmapay), outbuf);
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/* increase count (in ts) of data pushed to basertppayload */
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rtppcmapay->first_ts +=
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gst_util_uint64_scale_int (payload_len, GST_SECOND, clock_rate);
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/* store amount of unpushed data (in ts) */
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rtppcmapay->duration =
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gst_util_uint64_scale_int (avail, GST_SECOND, clock_rate);
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}
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return ret;
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}
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static GstFlowReturn
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gst_rtp_pcma_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpPmcaPay *rtppcmapay;
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guint size, packet_len, avail;
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GstFlowReturn ret;
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GstClockTime duration;
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guint32 clock_rate;
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rtppcmapay = GST_RTP_PCMA_PAY (basepayload);
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clock_rate = basepayload->clock_rate;
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size = GST_BUFFER_SIZE (buffer);
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duration = gst_util_uint64_scale_int (size, GST_SECOND, clock_rate);
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avail = gst_adapter_available (rtppcmapay->adapter);
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if (avail == 0) {
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rtppcmapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtppcmapay->duration = 0;
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}
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/* get packet length of data and see if we exceeded MTU. */
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packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
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/* if this buffer is going to overflow the packet, flush what we
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* have. */
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if (gst_basertppayload_is_filled (basepayload,
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packet_len, rtppcmapay->duration + duration)) {
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ret = gst_rtp_pcma_pay_flush (rtppcmapay, clock_rate);
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/* note: first_ts and duration updated in ...pay_flush() */
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} else {
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ret = GST_FLOW_OK;
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}
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gst_adapter_push (rtppcmapay->adapter, buffer);
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rtppcmapay->duration += duration;
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return ret;
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}
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gboolean
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gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin)
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{
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@ -17,8 +17,7 @@
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#define __GST_RTP_PCMA_PAY_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstbasertppayload.h>
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#include <gst/base/gstadapter.h>
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#include <gst/rtp/gstbasertpaudiopayload.h>
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G_BEGIN_DECLS
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@ -38,16 +37,12 @@ typedef struct _GstRtpPmcaPayClass GstRtpPmcaPayClass;
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struct _GstRtpPmcaPay
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{
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GstBaseRTPPayload payload;
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GstAdapter *adapter;
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GstClockTime first_ts;
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GstClockTime duration;
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GstBaseRTPAudioPayload audiopayload;
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};
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struct _GstRtpPmcaPayClass
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{
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GstBaseRTPPayloadClass parent_class;
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GstBaseRTPAudioPayloadClass parent_class;
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};
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gboolean gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin);
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@ -59,17 +59,9 @@ static GstStaticPadTemplate gst_rtp_pcmu_pay_src_template =
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static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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static void gst_rtp_pcmu_pay_finalize (GObject * object);
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GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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/* The lower limit for number of octet to put in one packet
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* (clock-rate=8000, octet-per-sample=1). The default 80 is equal
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* to to 10msec (see RFC3551) */
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#define GST_RTP_PCMU_MIN_PTIME_OCTETS 80
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GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPAudioPayload,
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GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
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static void
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gst_rtp_pcmu_pay_base_init (gpointer klass)
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@ -95,30 +87,24 @@ gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_rtp_pcmu_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_pcmu_pay_handle_buffer;
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}
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static void
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gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass)
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{
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rtppcmupay->adapter = gst_adapter_new ();
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GstBaseRTPAudioPayload *basertpaudiopayload;
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basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmupay);
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GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
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}
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static void
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gst_rtp_pcmu_pay_finalize (GObject * object)
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{
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GstRtpPcmuPay *rtppcmupay;
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/* tell basertpaudiopayload that this is a sample based codec */
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gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
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rtppcmupay = GST_RTP_PCMU_PAY (object);
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g_object_unref (rtppcmupay->adapter);
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rtppcmupay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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/* octet-per-sample is 1 for PCM */
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gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1);
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}
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static gboolean
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@ -132,114 +118,6 @@ gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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return TRUE;
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}
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static GstFlowReturn
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gst_rtp_pcmu_pay_flush (GstRtpPcmuPay * rtppcmupay, guint32 clock_rate)
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{
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guint avail;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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guint maxptime_octets = G_MAXUINT;
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guint minptime_octets = GST_RTP_PCMU_MIN_PTIME_OCTETS;
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if (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime > 0) {
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/* calculate octet count with:
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maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
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maxptime_octets =
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gst_util_uint64_scale_int (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime,
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clock_rate, GST_SECOND);
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}
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/* the data available in the adapter is either smaller
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* than the MTU or bigger. In the case it is smaller, the complete
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* adapter contents can be put in one packet. */
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avail = gst_adapter_available (rtppcmupay->adapter);
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ret = GST_FLOW_OK;
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while (avail >= minptime_octets) {
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guint8 *payload;
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guint8 *data;
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guint payload_len;
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guint packet_len;
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/* fill one MTU or all available bytes */
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payload_len =
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MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmupay), maxptime_octets),
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avail);
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/* this will be the total lenght of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
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/* create buffer to hold the payload */
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outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
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/* copy payload */
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gst_rtp_buffer_set_payload_type (outbuf,
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GST_BASE_RTP_PAYLOAD_PT (rtppcmupay));
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payload = gst_rtp_buffer_get_payload (outbuf);
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data = (guint8 *) gst_adapter_peek (rtppcmupay->adapter, payload_len);
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memcpy (payload, data, payload_len);
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gst_adapter_flush (rtppcmupay->adapter, payload_len);
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avail -= payload_len;
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GST_BUFFER_TIMESTAMP (outbuf) = rtppcmupay->first_ts;
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmupay), outbuf);
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/* increase count (in ts) of data pushed to basertppayload */
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rtppcmupay->first_ts +=
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gst_util_uint64_scale_int (payload_len, GST_SECOND, clock_rate);
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/* store amount of unpushed data (in ts) */
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rtppcmupay->duration =
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gst_util_uint64_scale_int (avail, GST_SECOND, clock_rate);
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}
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return ret;
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}
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static GstFlowReturn
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gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpPcmuPay *rtppcmupay;
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guint size, packet_len, avail;
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GstFlowReturn ret;
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GstClockTime duration;
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guint32 clock_rate;
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rtppcmupay = GST_RTP_PCMU_PAY (basepayload);
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clock_rate = basepayload->clock_rate;
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size = GST_BUFFER_SIZE (buffer);
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duration = gst_util_uint64_scale_int (size, GST_SECOND, clock_rate);
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avail = gst_adapter_available (rtppcmupay->adapter);
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if (avail == 0) {
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rtppcmupay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtppcmupay->duration = 0;
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}
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/* get packet length of data and see if we exceeded MTU. */
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packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
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/* if this buffer is going to overflow the packet, flush what we
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* have. */
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if (gst_basertppayload_is_filled (basepayload,
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packet_len, rtppcmupay->duration + duration)) {
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/* note: first_ts and duration updated in ...pay_flush() */
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ret = gst_rtp_pcmu_pay_flush (rtppcmupay, clock_rate);
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} else {
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ret = GST_FLOW_OK;
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}
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gst_adapter_push (rtppcmupay->adapter, buffer);
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rtppcmupay->duration += duration;
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return ret;
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}
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gboolean
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gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)
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{
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@ -17,8 +17,7 @@
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#define __GST_RTP_PCMU_PAY_H__
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#include <gst/gst.h>
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#include <gst/rtp/gstbasertppayload.h>
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#include <gst/base/gstadapter.h>
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#include <gst/rtp/gstbasertpaudiopayload.h>
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G_BEGIN_DECLS
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@ -38,16 +37,12 @@ typedef struct _GstRtpPcmuPayClass GstRtpPcmuPayClass;
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struct _GstRtpPcmuPay
|
||||
{
|
||||
GstBaseRTPPayload payload;
|
||||
GstAdapter *adapter;
|
||||
|
||||
GstClockTime first_ts;
|
||||
GstClockTime duration;
|
||||
GstBaseRTPAudioPayload audiopayload;
|
||||
};
|
||||
|
||||
struct _GstRtpPcmuPayClass
|
||||
{
|
||||
GstBaseRTPPayloadClass parent_class;
|
||||
GstBaseRTPAudioPayloadClass parent_class;
|
||||
};
|
||||
|
||||
gboolean gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin);
|
||||
|
|
|
|||
Loading…
Reference in a new issue